CompUSA and Xandros announced a partnership today. Xandros, a provider of easy-to-use Linux alternatives to Windows, announced today that the retail giant, CompUSA, is now selling the Xandros Desktop product in all of its stores, nationwide. The Xandros product line offers home and business users a complete, stable, and secure alternative to costly Windows systems. I've been thinking about trying Xandros myself on my home lab setup.
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Cisco at 12am March 6th will launch their new Unified Communications System (www.cisco.com/go/unified) aimed at streamlining business processes, and helping to drive productivity. Unified Communications (UC) will feature new presence, desktop tools, mobile integration and network intelligence to improve business agility and customer interaction, but just as importantly as I will indicate later, Cisco is fully embracing the SIP standard on their desktop phones. I interviewed Cisco last week and they told me that I was the first journalist or analyst to have a "first look" at this major announcement.
Cisco IP phone & the new IP Communicator softclient
Perhaps I misread the tone of the Cisco representatives during my call, but they initially seemed to downplay the significance of Cisco's embracing of SIP in favor of focusing on the entire Unified Communications platform. But in my opinion, Cisco embracing SIP is just as big news as their Unified Communications system, but more on that later.
Essentially, Cisco Systems, unveiled the Cisco Unified Communications system, which is a new suite of voice, data and video products and applications specifically designed to help organizations of all sizes to communicate more effectively. Cisco Unified Communications replaces AVVID and IP Communications. The system will allow customers to integrate their communications system with their IT infrastructure, streamlining business processes for the way effective businesses need to work today.
Based on the Cisco Service-Oriented Network Architecture (SONA) announced in December 2005, the Cisco Unified Communications system is an open and extensible platform for real-time communications based on presence, mobility and the intelligent information network. It uses the IT data network as the service delivery platform helping workers to reach the right resource the first time by delivering presence and preference information to an organization's employees.
But we're not done - Cisco is also launching a major revision of CallManager, namely CallManager 5.0, a core piece of their VoIP platform. As part of this major version release, Cisco now is supporting SIP on CallManager which in effect will enable 3rd party SIP phones to communicate with CallManager. This is a huge deal since in the past you were forced to buy Cisco phone running the Skinny protocol. Now you have a choice of IP phones. Cisco believes that their phones with tight integration with applications and strong features will continue to help them keep their market leading position.
Vicky McGovern Director of Marketing IP Communications Group said, "We believe that this changes everything. This is our new mantra, 'this changes everything' introducing Cisco's Unified Communications. And we really believe that this changes everything for three primary constituencies."
Side Note: "This changes everything" was actually a quote used by Jeff Bezos given to the Wall Street Journal after viewing a hush-hush preview for “Ginger,” an unbelievably-hyped launch for what turned out to gyroscopic scooters called the Segway. Guess Mr. Bezos should have trademarked/patented the term and then "pulled an NTP" and extorted Cisco for $612.5 million to use the phrase.
Vicky continued, "This changes everything first and foremost for our end-user customers. We believe this changes everything because we believe there's a key business imperative that business decision makers are not addressing in their environments today. And that is that communications is absolutely critical to business processes and communications has been looked at as a "silo application" deployed out of necessity versus being views as a key business asset. A lot of the research that Cisco has done points to the fact that there is a tremendous impact to business processes due to the lack of effective communications. Unified communications can help in generating that effectiveness and eliminating the bottlenecks and enabling businesses to grow."
Vicky explained the second constituencies will change everything for their partners due to a plethora of new partner tools. "We will work with our partners to articulate 'this changes everything' message to the key business decision maker. Cisco is well-known for being in with the technical decision maker and speaking their language. Now you're going to see us address the business decision maker with tools that will enable them to speak that language to speak to the business decision maker. We will offer sales tools, configuration tools, and pricing models and packaging models and support services that they can augment their kit bag with - so a whole new way of Cisco working with our channel partners bring this to market."
The third change is for Cisco themselves. They explained that they are repositioning their entire IP Communications (IPC) portfolio, moving it beyond IPC into the unified communications realm. Again, Cisco will be reaching out to a new audience - the business decision maker and even the voice decision maker - to help them understand how these new developments in technologies can help them solve business issues they may not even know they have. Cisco stated, "we are embarking on a new era - broadening our image from “the leader in switching & routing” to one that enables and embraces all forms of communication". 
Unified Communications System Architecture
"The Cisco Unified Communications system is the first true second-generation Internet Protocol (IP) Communications system providing not just telephone services, but rather a rich communications environment that seamlessly integrates voice, video and data collaboration in one system," said Charles Giancarlo, chief development officer, Cisco Systems, Inc.
The Cisco Unified Communications system is based on Cisco's IP Communications portfolio including Cisco CallManager, Cisco Unity, Cisco MeetingPlace and Cisco IP Contact Center and now includes additional innovative products, applications, features and capabilities. New to the Cisco Unified Communications system are Cisco Unified Personal Communicator, Cisco Unified Presence Server and Customer Interaction Analyzer. Current customers will be able to upgrade their existing systems to take advantage of the new capabilities.
"Miercom has exercised and reviewed key components of the entire Cisco Unified Communications system and after seeing it in action, we believe that Cisco has leapfrogged their competition in a number of areas," said Ed Mier, principal, Miercom. "Cisco's native implementation of SIP, which is interoperable with Skinny Call Control Protocol (SCCP) helps give customers investment protection for their system so that it can adapt as quickly as the standard does."
Barry O'Sullivan vice president and general manager of Cisco's IP Communications Business Unit told me, "Our strategy is to put as much intelligence onto the network to allow applications whether our applications or others applications to take advantage of that intelligence. So call processing intelligence, presence intelligence, and rich-media applications. Our strategy is to be open and extensible. In other words, we have embraced SIP and built in natively into our platform and we'll make these capabilities available to third party applications and phones as well as our own applications and phones".
When asked, "Any plans to federate", Barry replied, "Yes we have implemented SIP and SIMPLE into our Presence Server and we'll make that available to anyone that wants to federate with us."
Barry explained, "Building native SIP Support for Cisco Unified CallManager, Unified CallManager Express and Unified SRST we feel is a key differentiator for us. Because we built our system from the ground up to be an IP system we can do this. CallManager supports SIP natively, and SRST and CallManager Express support SIP natively in IOS - those are pieces of software that run under IOS." He continued, "Our competitors on the other hand typically have a separate system that supports SIP and proxy across to their traditional PBX." He explained that this makes it difficult for customers to manage two systems and provisioning users across two systems without feature transparency across the two systems where as the Cisco system does feature transparency.
Up to now Cisco CallManager has only been available on an "open server" running Windows. With this launch, Cisco is announcing is that they will now offer customers an appliance, based on the Linux operating system. This is another huge piece of news. CallManager on Linux? Wow! I still remember installing CallManager 1.0 many eons ago in TMC Labs. I forget now if I installed it on Windows NT 4.0 or Windows 2000. In any case, IT departments that were hesitant to run CallManager on Windows, or are Linux shops, will be excited to hear this news.
Another new product as part of this launch is Cisco Unified Personal Communicator, a software client which is a single portal into all your communications applications handling voice, video, IM, unified messaging, and collaboration. Barry, "Unified Personal Communicator gives you a single unified interface to all of these. It's your one software communications device on your desktop so you can see your voicemails, you can enter into a collaboration session, and our vision is that you will be able to drag-and-drop your presentations into Personal Communicator, drag-and-drop video to share video, etc. It will be the most visible thing at the user desktop." Cisco also mentioned they are working closely with Microsoft and will offer integration with Microsoft Office Communicator if customers wish to use that software client instead. I don't have a screenshot yet for Unified Personal Communicator but will update this post if I get one.
Cisco next explained "Cisco Customer Interaction Analyzer" for the contact center. Barry claimed, "This technology can listen to a call and using word spotting, looking at tone, inflection, and cadence to figure out the quality of the interaction and make an analysis. So it can figure out if the customer is happy or angry or was concerned about a problem and then make that analytic information available from the network to applications so you can do intelligence call routing, train your customer service representatives to be better, give that customer better service next time they call." I inquired whose technology they use and they said it is OEM'ed from eLoyalty.
Tom: So you are offering a new firmware download to enable SIP on your phones. Does this mean you are abandoning Skinny (SCCP)?
Barry: No, some of our customers will want to continue with Skinny and some of our customers will want SIP. We're going to continue to provide choice for them. But we really think the market is going to go to SIP pretty quickly and that's why we are providing full SIP functionality on our SIP phones
Tom: Is there 1-to-1 feature functionality between the new SIP firmware and the Skinny firmware? Is there anything missing?
Barry: It's over 90% is what we're saying - except for some obscure features. There is feature transparency between a SIP phone and Skinny phone, so if you are in transition with some SIP and some Skinny phones. For example you can do a shared line between a SIP phone and a Skinny phone or park a call on a SIP phone and pick it up on a Skinny phone. So we have full feature transparency. We can do this again because we support SIP natively but our competitors only support this with a bolt-on server, so it is much more difficult to have this feature transparency.
One really cool new feature is announcing is that Cisco will support dual-mode phones through partnerships with Nokia and RIM (Blackberry 7270). Good thing RIM settled with NTP or this would be a very short partnership! Cisco worked with Nokia and RIM to offer a softphone client on their dual-mode phones so you can get full-featured access to CallManager when you are within range of your wireless WiFi network.
One interesting fact Barry pointed out was, "We've seen a huge acceleration and adoption of IP communications. We're displacing 12,000 of our competitors' phones every business day. We shipped 7.5 million phones and it took us 3 years to ship the first million and just 3 months to ship the most recent million."
Tom: Was SIP a major driving force behind enabling Cisco's Unified Communications?
Barry: Yeah, it's SIP everywhere pretty much. Instead of managing voice and video and IM as separate things, what SIP lets you do is manage them all in one session. So you can seamlessly escalate IM to a voice session to a session call. The concept a call gets broadened to the concept of a session and that is much easier to manage from a software point of view.
Tom: <thinking in my head> Well, they don't call it Session Initiation Protocol (SIP) for nothing!
Barry explained the power of SIP quite succinctly and accurately even if I already knew this.
Here is an overview/summary of the Cisco Unified Communications new products & features.
- Cisco Unified Personal Communicator simplifies the way workers share information by helping them to communicate in real time. Its user-friendly GUI (Graphical User Interface) makes it easy to move through multiple communications applications. The Unified Personal Communicator bridges the gap between the stand-alone applications on the desktop, telephone and network. Using dynamic presence information, employees can search existing directories to locate contacts and simply "click to call" using voice and video, allowing them to exchange ideas face-to-face. The virtual nature of IP networks allows remote or traveling employees to securely access these tools from wherever they are.
- The Cisco Unified Presence Server collects information about a user's status, such as whether or not they are using a device such as a telephone, personal computer or video terminal at a particular time. Using this information, applications such as Cisco Unified Personal Communicator and Cisco Unified CallManager can help users connect with colleagues more efficiently by determining the most effective method of communication. The Cisco Unified Presence Server aggregates presence information from the network as well as Cisco Unified CallManager and third-party devices using SIP and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) and then publishes that information to Cisco Unified IP Phones, Cisco Personal Communicator and third-party services and applications such as IBM Lotus Sametime and Microsoft Live Communications Server (LCS) 2005.
- The Customer Interaction Analyzer is being introduced to maximize effective communications with customers, a new approach to analytics in the contact center. It uses information from customer interactions, including self service and agent assisted interactions, to determine things like customer distress, agent distress, silence and word patterns. The data helps to give the conversations business context and can help a business to coach and train agents, make changes to processes and self service scripts based upon findings - ultimately creating better customer relationships and growth for the business. Additional new features of the Cisco Unified Communications System include the following:
- Cisco Unified CallManager 5.0 and Cisco Unified CallManager Express 3.4 and Survivable Remote Site Telephony (SRST) 3.4 now natively support SIP, effectively opening up the system to an emerging standards-based developer community while retaining the current security and resiliency features. A new program, SIP Verified, provides third-party verification for voice, data and video SIP endpoints. An initial set of vendors who have completed this testing is also announced.
- Cisco Unified CallManager 5.0 is now available in a choice of operating models based on customer and channel partner preference. A new appliance model version based on Linux is available now and a version based on the existing open operating system model is scheduled to be available within 12 months.
The big knock against Cisco has been their proprietary Skinny protocol, now that Cisco fully supports SIP their competitors will no longer have that ammunition. Of course, they'll probably harp on the fact that Cisco's IOS operating system is proprietary and closed. You never hear people asking Nortel or Avaya to make their PBX operating system code publicly available or "open source". In any case, it's a bright day for the VoIP industry when an industry heavyweight such as Cisco puts their full weight behind the SIP standard.
Asterikast is a podcast that teaches and explains about the VoIP-capable Asterisk PBX by Digium. According to Asterikast, "We also plan on having videos that can help step you through the process of setting up your very own Asterisk PBX."
They already have two episodes available for download. Episode I, "the very first episode of Asterikast" they cover how to compile Asterisk and setup one SIP phone. Episode II has much higher quality video, audio and on-screen footage. They cover voicemail, conference bridges, macros in Asterisk and TDM/TDM cards. Looks like they're also offering a computer pre-installed and pre-configured with Slackware and Asterisk for $800.00 if your a bit intimidated to install Linux + Asterisk. Both are actually pretty easy to install, and in fact it's worth checking out the videos just to see how easy it truly is.
The Philips VP5500 WiFi VoIP phone was announced way back in September 2005 and it was finally launched today in the Netherlands of all places. No offense Netherlands, but how come you get first dibs on this cool phone?
Anyway, the sleekly styled VP-5500 is powered by Linux and lets users enjoy live video calls using its built-in VGA camera (640x480 resolution) that rotates up to 240 degrees and supports 30 FPS. Video calls are displayed on a 2.2" color LCD supporting 64k colors. The VP5500 features a video out port that lets others watch the video on a TV. Though hooking up a video wire kind defeats the purpose of using a wireless videophone, don't ya think?
You can also zoom in on captured still images stored on the phone's internal 1MB memory. It also features a built-in speakerphone and hands-free headset compatibility.
As previously mentioned, the Philips VP-5500 VoIP Videophone runs on Linux, so they've built this phone around "standards", such as Wi-Fi, WPA, and most importantly the SIP protocol standard. The VP5500 can be upgraded wirelessly and will support applications developed by service providers. No date has been set for a release outside of Holland, however Philips is looking to partner with third party operators.
Tello was launched today by Pulver with some help from Craig McCaw, telecom banker Michael Price and former Apple CEO John Sculley. What is Tello? Tello enables enterprise users to see the presence of the person they are trying to reach - whether the users is on a phone, cellphone, etc., regardless of which IM client service they use, i.e. MSN Messenger, AOL/AIM, Yahoo! Messenger, etc. (Note: only SIP-based IM services are supported as far as I can tell.)
The concept is simply to "bridge" the various IM services and be able to see the presence of the user. It features IP-PBX integration with the likes of Avaya and even the popular Asterisk open-source Linux-based IP-PBX, as well as a Blackberry client. Ironically, I just blogged about IM interoperability between AOL & IBM's Sametime and I predicted IM interoperability would finally happen in my 2006 predictions. Thanks for proving me right Mr. Pulver!
IP Democracy has some thoughts on this news worth a look, as does BusinessWeek, Om, SiliconBeat, and the WSJ. Mostly positive reviews from what I read. Also, according to Tello, "The solution is delivered as a hosted Instant Communications and Collaboration network service, Tello Connect, and is available with complementary Desktop and Mobile Client applications. It provides a Real Time Communications Presence & Connectivity Hub that allows people to instantly locate, contact, and connect with friends, colleagues, and partners using the devices and applications that they already know. Using the Tello Desktop and mobile applications, individuals can, at a glance, easily determine the availability of their contacts at any time anywhere in the world and initiate rich media multi-modal communications with the click of a button."
Here's how it works by leveraging the SIP standard. Tello's VoIP federation service is built around a directory which provides SIP routing based on dialed telephone number or SIP URI. When a call is dialed (either via a SIP UA or an IP-PBX), the dialed call is signaled using SIP to Tello Connect which checks several databases and applies a set of customer-defined policies to determine the optimal SIP route. Thus, using this solution which queries multiple IM databases, you can increase the percentage of calls that remain all IP and thereby save money.
If this sounds eerily similar to FWD, the Free World Dialup service Pulver created, a SIP-based direct SIP URI dialing directory, you would be correct. In fact, when I read this news on a few news sites, I thought to myself "Didn't Jeff already try this with FWD?" The added benefit with Tello is that it performs "hosted federation" and integration with just about any IM service that supports the SIP standard, which includes MSN Messenger, AOL, Yahoo Messenger, and several others. At first glance, this hosted solution to the IM interoperability dilemma seems like a great idea. However, it seems like a near-term solution to me. If the big IM players get their act together and decided to use the same SIP/SIMPLE IETF standard, which allows for presence sharing, you wouldn't even need Pulver's hosted solution. In addition, many of the IM players have announced federation plans already, including Google.
Honestly, I'm not convinced a hosted service provider that centralizes and consolidates all of your presence information is going to be a "killer app" that people will pay a monthly fee. The target for this service is the SMB market, not consumers, so it's possible that SMB CEOs looking to improve employee productivity will sign off on essentially "renting presence knowledge". It could especially help sales people quicky and easily reach their B2B sales prospects instead of playing voicemail tag or even email tag. Still, the verdict on IM in the enterprise being a productivity enhancer or productivity waster (chatting with your spouse or friends at work) is still out. I should mention this service won't work with Skype's proprietary chat mechanism, which is the #1 used VoIP application in the world today.
Also, I'm going to assume Tello's hosted service requires your various usernames & passwords to the various IM services in order to logon, authenticate and access your presence information. If that's the case, then Tello will also face an uphill battle against users that don't want to share their various IM client username and passwords. Considering MSN Messenger accounts use Microsoft's Passport which shares the same account as your hotmail.com email address, Hotmail users might be wary over sharing (with a third-party) their password to their email account containing personal and confidential emails. Am I wrong here?
Some news from PalmSource, about them joining a forum that is trying to "standardize" Linux components and APIs that run on mobile phone devices. The reason why Microsoft has been making such strong in-roads in the mobile phone market arena is because they have strict requirements and they provide a baseline operating system that makes it easier for developers to cross-develop applications for different, often competing Windows Mobile devices running Windows Mobile 5.0 (or earlier generations). Here's the release from their website.
PalmSource, Inc., provider of Palm OS, a leading operating system powering next generation phones and mobile devices, today announced that it is a founding member of the Linux Phone Standards (LiPS) Forum. The LiPS Forum, a consortium of leading companies, has come together to accelerate the adoption of Linux in fixed, mobile and converged devices by standardizing Linux-based services and APIs that most directly influence the development, deployment and interoperability of applications and user-level services. Alongside PalmSource, the founding members include France Telecom/Orange, FSM Labs, Huawei, Jaluna, MontaVista Software, MIZI Research, Open Plug, Arm, Cellon and Esmertec.
The primary goal of the LiPS Forum is to establish standards for the growing numbers of companies providing Linux-based technologies for mobile, fixed and converged telephony terminals. With the rapid increase of Linux's popularity in these markets, there is an increasing need for industry standards to avoid fragmentation and ensure interoperability of technologies from different vendors. The LiPS Forum intends to support device manufacturers and operators in bringing to market Linux-based devices at a lower cost, while facilitating the programming and development process for software and semiconductor vendors. Additionally, the LiPS Forum plans to foster communication between the open source community and the telecom industry in order to drive market awareness.
Michael Kelley, senior vice president of engineering from PalmSource commented, "Becoming a part of the LiPS Forum further demonstrates our belief in the potential of Linux and our plans to developing on Linux. We believe that by simplifying the adoption of Linux in fixed, mobile and converged devices, and working to ensure that they match the requirements of operators and consumers, the LiPS Forum will play an important part in making Linux a truly mass market proposition.
About the Linux Phone Standards Forum (LiPS)
The Linux Phone Standards (LiPS) Forum is a consortium of leading industry players formed to standardize the Linux-based services and Application Programming Interfaces (APIs) that most directly influence the development, deployment and interoperability of applications and user-level services.
The success of Linux in phones, as elsewhere, is reliant on standards that ensure interoperability and guard against fragmentation. To date, standardization efforts have focused on the important question of Linux kernel optimization to achieve improved boot time, power management, system footprint and other performance-related factors. But for mass-market telephony terminals, standards that enable key applications and services to be deployed with a high degree of interoperability are at least as important as performance characteristics. LiPS has been created to meet this need that is crucial for modern phones and thus complement current initiatives around Linux standardization.
The founding members are ARM, Cellon, Esmertec, France Telecom/Orange, FSM Labs, Huawei, Jaluna, MIZI Research, MontaVista Software, Open-Plug and PalmSource, Inc.
I had a conference call last week with Covergence, about their pending announcment with Vonage that explains that Covergence is the company that powers Vonage's e911 service. Covergence is an interesting company that has been stealthily flying under the VoIP radar screen, including my radar.
Basically, Covergence is similar to a Session Border Controller (SBC) but much more advanced offering unified security and management of network applications. For example, it is application aware, including SIP-aware and can route e911 PSAP information to the appropriate 911 emergency services gateway. Vonage will use Covergence's Eclipse to provide E-911 call routing, E-911 call recording and E-911 call detail records. Covergence told me, "We have very sophisticated SIP call routing and packet manipulation, we're a fully back-to-back terminating device, so we can make sure that we interject the right PSAP information and route the call. And because we do have this ability to stream all the media we can record all the calls, log all the calls, to provide confidence this is actually what happened on that (911) call." In a nutshell, the Covergence solution can intercept the SIP packets, determine it is a 911 call, perform SIP-level field manipulation and route the call to the appropriate center before hitting Vonage's call processing network infrastructure.
I actually wanted to work on a detailed writeup on Covergence over the weekend, but unfortunately, I was very busy this weekend finishing up some VoIP product reviews in preparation for being at Internet Telephony Expo next week. So instead, here's my (relatively) quick take on Covergence...
Covergence raised over $16 million and has over half of that available and is funded by North Bridge Venture Partner and Highland Capital Parners, too very reputable firms. Covergence told me, "We see Eclipse as the first unifed SIP-based protocol for security and management solutions and what we've done is architect a real-time communications system that combines VoIP and collaboration software applications for deployment both in the enterprise and the carrier ."
They told me that they use a hardened version of SUSE Linux for good reliability, security, and scalability. Covergence offers a scalable family of hardened appliances providing policy-driven, application-level security, control, monitoring and interoperability. It can even perform H.323-to-SIP transcoding. Even more interesting is that they can transcode between IBM's Sametime and Microsoft's LCS (Live Communications Server) to provide interoperability between the entrenched SameTime application and the up-and-coming LCS platform.
Also, as part of their solution that offer an encrypted, authenticated, and validated connection to the subscriber to ensure subscriber confidentiality and integrity. I asked them whether or not existing CPE (customer premise equipment) such as ATAs (analog telephone adaptors), such as the Cisco ATA-186, would support a firmware update to support encryption of the VoIP packets and if the current crop of ATAs had the processing horsepower to handle encryption. They said all the ATA manufacturers are working very hard to provide SRTP media encryption in their future ATA devices, and that it really depended on how new the chipset on the ATA is and if it would support encryption via a simple firmware upgrade. I guess I won't be using Ethereal or other packet sniffer software on my home Vonage line to eavesdrop and record my wife's conversations.
Just kidding! They also said that they offer intrusion and attack prevention protection at the service provider network.
I asked them "So how much latency do you add to the network, especially with encryption added on the fly?" They responded, "Well, that's part of the magic of what we've done because we special purposed architected our box to maintain low latency, so we're in the 20 microsecond latency area for full throughput calls."
Here are some highlighted features of the Covergence solution:
Hardware Based Media Processing
- Custom-designed, hardware-based media processing cards for environments with heavyreal-time media (e.g. audio, video) traffic
- Dual Motorola Freescale processors with integrated crypto engines, 24 Gb/s QOS-aware switch fabric, proprietary custom logic
- Media switching, media NAT, SRTP media encryption, media validation, media replication, QOS monitoring, CODEC translation
Security
- Cryptographic authentication
- Signaling and media encryption
- Stateful signaling and media validation
- Denial-of-service attack mitigation
- Intrusion prevention
- Virus scanning
- Malicious URL filtering
Control
- Signaling and media control
- SIP-aware NAT traversal
- Quality of service (QOS) control
- Identity-based access control
- File transfer control
- Instant message content control
- URL access control
Monitoring
- Session detail recording
- QOS detail recording
- Instant message recording
- Voice and video recording
- File transfer recording
- System and administrative event logging
In any event, getting back to the Vonage news, using Eclipse, Vonage can intelligently route E-911 calls to the correct emergency services gateway, based on the customer information retrieved from the E-911 service bureau database. This will assist Vonage in its aggressive plans to meet FCC guidelines required of VoIP service providers to offer reliable E-911 services. In addition, Eclipse allows Vonage to record and create records of E-911 calls, helping the company to easily and accurately view historical information, and ensuring that the Vonage E-911 environment continually delivers optimum call quality.
Here's a snippet of quotes taken from today's news release and a screenshot of the Covergence Java interface:
"Vonage continues to seek out those technologies that will ensure our customers' safety and is rolling out E-911 services that will meet or exceed those currently available through traditional phone service," said Martin Hakim Din, Senior Vice President, Vonage Network. "Covergence provides us with another key piece of the puzzle and will help us to ensure that our customers receive reliable and efficient emergency phone service."
"Broadband phone service is becoming a viable replacement for traditional phone service and Covergence is dedicated to ensuring that pioneer providers like Vonage can provide secure, reliable and quality residential services," said Bob O'Neil, president and CEO, Covergence. "We're proud to be supporting Vonage's plans as they rapidly augment and optimize their E-911 environment to give their growing subscriber base the added assurance that emergency functions delivered through broadband will exceed their expectations."
There's a new book on the popular open-source Asterisk IP-PBX phone system out, titled Asterisk : The Future of Telephony that you may be interested in. One of the coauthors stated, "Shortly after discovering Asterisk, I realized that this phenomenon was going to radically alter the telecommunications industry. I knew that open source telephony represented a bright new future: not just for me, but also for the telecom industry as a whole," says Jim Van Meggelen, "Asterisk is as much a cultural revolution in the IT and telecom industries as it is a technical one." I need to get my hands on a copy and do a book review. In meantime, here are some other thoughts from the various book authors...
Internet telephony with VoIP (Voice over Internet Protocol) hasn't yet reached critical mass, but it's poised to. VoIP promises huge cost savings, but its ability to move data, images, and voice traffic over the same connection will undoubtedly cement its place in the future of telecommunications. That's why so many IT administrators and developers are exploring VoIP-based private telephone switching systems within the enterprise. The efficiency that network users can reach with it is almost mind-boggling. And cheap, if the system is built with open source software PBX like Asterisk.
There are commercial VoIP options out there, but many are expensive systems that run old, complicated code on obsolete hardware. Asterisk runs on Linux and can interoperate with almost all standards-based telephony equipment. Asterisk embraces the concept of standards-compliance, but also gives users the freedom to choose how to implement their systems.
"Asterisk is arguably the most influential and exciting piece of software since the operating system it runs on--Linux," says coauthor Leif Madsen.
"Asterisk--or at least the open source telephony system concept--is going to change the telecommunications industry in a dramatic way, but its learning curve can certainly be a barrier. This book is designed to lower the barrier of entry, allowing the software to proliferate into the world, and to dramatically change the telecommunications world as we know it. All the big players have it in their labs, and have for some time now. They all know that this is the future of telephony."
"I believe this book will allow people to catch the vision of just how powerful and flexible Asterisk is as a telephony platform," says Jared Smith, the third coauthor of the book. "Finally, geeks can fiddle with their phone calls just like mechanics fiddle with their cars. They can add features, increase performance, add redundancy, and increase collaboration."
"It should also give IT managers an understanding of why their geeks are suddenly so excited about the phone system," adds Van Meggelen.
With Asterisk, users are no longer dependent on expensive and inflexible systems that are tuned to the vendor's needs, rather than the end user's.
Asterisk's flexibility comes at a price, however: it's not a simple system to learn, and the documentation has hitherto been lacking. Linux pros need to learn a bit about telephony; telecom pros need to learn a bit about Linux. "To my fellow telecom professionals I say 'learn Asterisk--it's going to transform our industry, and you're gonna love it because you get to look your customer in the eye and say yes a lot,'" says Van Meggelen.
With "Asterisk: The Future of Telephony," the future is no longer unmapped. The book will help readers to truly understand the core concepts of Asterisk. "The software really isn't that complex once you have an understanding of the main concepts, but those concepts can at times can seem disparate and unwieldy," Madsen notes. "This book will give you the grounding and knowledge required to explore the more complex concepts, which would otherwise be impossible."
This new book offers a complete roadmap for installing, configuring, and integrating Asterisk with existing phone systems. It walks readers through a basic dial plan step by step, and gives them enough working knowledge to set up a simple but complete system. The book outlines all the options, and shows how to set up voicemail services, call conferencing, interactive voice response, call waiting, caller ID, and more. Readers will also learn how Asterisk merges voice and data traffic seamlessly across disparate networks. And they won't need additional hardware: for interconnection with digital and analog telephone equipment, Asterisk supports a number of hardware devices.
The future of telephony is bright--and with "Asterisk: The Future of Telephony," you can be ready for it.
Additional Resources:
Chapter 5, "Dialplan Basics," is available online at: http://www.oreilly.com/catalog/asterisk/chapter/index.html
For more information about the book, including table of contents, index, author bios, and samples, see: http://www.oreilly.com/catalog/asterisk/index.html
Linux.com took a look at four Linux VoIP softphone clients and did some comparative analysis. The Linux softphones included Kiax, Linphone, Twinkle, and CounterPath's X-Lite. Three of the four softphones use the standard SIP protocol with Kiax being the one exception - it uses Asterisk's IAX protocol. Check out the full review of the Linux softphones.
Just some quick news to share about Artesyn Communication Products which is partnering with MontaVista to offer Linux Carrier Grade Edition (CGE) for its telecom blades and modules. Embedded Linux is becoming more imporantant in the telecom sector and not just Asterisk enterprise solutions, but in the carrier sector as well.
Artesyn will offer bundled, certified MontaVista CGE solutions for its PICMG 2.16, AdvancedTCA, AdvancedMC, ProcessorPMC blades and modules, preinstalled in flash memory. Artesyn blades and modules equipped with MontaVista CGE provide a modular open architecture platform for building scaleable, high-availability network infrastructure equipment.
“Embedded Linux is emerging as a dominant platform for building high-availability network infrastructure products, and we are committed to offering our telecom equipment OEM customers the finest carrier grade Linux solutions,” said Todd Wynia, vice president of marketing at Artesyn. “Soon, Network Equipment Providers (NEPs) will be able to purchase MontaVista’s carrier grade Linux directly from Artesyn, certified by MontaVista, and pre-installed in flash on our telecom blades and modules.”
MontaVista Linux CGE is an open and flexible development platform designed specifically to address the unique and demanding requirements of carrier grade class applications, with a strong focus on open standards and high availability services. MontaVista CGE is the industry’s most field tested and proven Linux distribution. It is the only carrier grade Linux deployed by the world’s leading NEPs in carrier networks for over three generations of product, and it serves as the foundation for advanced telecom solutions currently in design. MontaVista CGE is also the only carrier grade Linux operating system that complies with both the Service Availability Forum (SA Forum) Application Interface Specifications for high availability services and the Open Source Development Lab’s (OSDL) requirements for Carrier Grade Linux 2.0.
“As the telecom industry responds to the extreme pressure to redesign their networks and introduce new advanced capabilities, NEPs are requiring a reliable, robust and flexible software platform to build out sophisticated designs while not putting their project schedules at risk,” said Peder Ulander, vice president of marketing at MontaVista. “Artesyn has been a pioneer in the development of open architecture telecom platforms and recognizes the need for the telecom industry to leverage cost-effective off-the-shelf (COTS) components with advanced Linux technologies. MontaVista atop Artesyn’s blades and modules gives our joint customers a tested, out of the box, best of breed foundation for building and deploying reliable, feature-rich, carrier-class solutions.”
MontaVista CGE is available immediately from MontaVista for Artesyn’s PICMG 2.16 Katana 750i, 752i, 3750 and 3752 blades, which feature PowerPC processing complexes, high-performance packet-switched backplane interfaces, PMC/PTMC mezzanine expansion, and integrated IPMI system management. MontaVista CGE is also available for Artesyn’s PowerQUICC-based Pm8560 protocol engine, an octal E1/T1 card optimized for SS7/SIGTRAN signaling. Artesyn will announce bundled, certified MontaVista CGE solutions for select PICMG 2.16, AdvancedTCA, AdvancedMC and ProcessorPMC blades and modules at a future date.
“We’re excited to see Artesyn and MontaVista join forces, and look forward to working with them on future PowerPC- and PowerQUICC processor-based high-availability Linux platforms,” said Jeff Timbs, a marketing director at Freescale Semiconductor's Networking and Computing Systems Group. “Artesyn blades running MontaVista CGE provide an excellent open architecture platform for developing and deploying telecom infrastructure applications targeting Freescale processors.”